]> git.sesse.net Git - mlt/blobdiff - src/modules/avformat/consumer_avformat.c
Fix muxing vorbis into webm/mkv with libavcodec < 55.
[mlt] / src / modules / avformat / consumer_avformat.c
index 6a4982c3b9293b5f6ec309d80c6396b10eb436cf..33b3a93f983e05597ae0830be5ae0503cfc41924 100644 (file)
@@ -1178,7 +1178,7 @@ static void *consumer_thread( void *arg )
        int count = 0;
 
        // Allocate the context
-       AVFormatContext *oc = avformat_alloc_context( );
+       AVFormatContext *oc = NULL;
 
        // Streams
        AVStream *video_st = NULL;
@@ -1232,6 +1232,22 @@ static void *consumer_thread( void *arg )
        if ( filename == NULL || !strcmp( filename, "" ) )
                filename = "pipe:";
 
+#if LIBAVUTIL_VERSION_INT >= ((53<<16)+(2<<8)+0)
+       avformat_alloc_output_context2( &oc, fmt, format, filename );
+#else
+       oc = avformat_alloc_context( );
+       oc->oformat = fmt;
+       snprintf( oc->filename, sizeof(oc->filename), "%s", filename );
+
+       if ( oc->oformat && oc->oformat->priv_class && !oc->priv_data && oc->oformat->priv_data_size ) {
+               oc->priv_data = av_mallocz( oc->oformat->priv_data_size );
+               if ( oc->priv_data ) {
+                       *(const AVClass**)oc->priv_data = oc->oformat->priv_class;
+                       av_opt_set_defaults( oc->priv_data );
+               }
+       }
+#endif
+
        // Get the codec ids selected
        audio_codec_id = fmt->audio_codec;
        video_codec_id = fmt->video_codec;
@@ -1310,9 +1326,6 @@ static void *consumer_thread( void *arg )
                }
        }
 
-       oc->oformat = fmt;
-       snprintf( oc->filename, sizeof(oc->filename), "%s", filename );
-
        // Get a frame now, so we can set some AVOptions from properties.
        frame = mlt_consumer_rt_frame( consumer );
 
@@ -1615,9 +1628,11 @@ static void *consumer_thread( void *arg )
 #endif
 #if LIBAVCODEC_VERSION_MAJOR >= 54
                                                        audio_avframe->nb_samples = FFMAX( samples, audio_input_nb_samples );
+#if LIBAVCODEC_VERSION_MAJOR >= 55
                                                        if ( audio_codec_id == AV_CODEC_ID_VORBIS )
                                                                audio_avframe->pts = synth_audio_pts;
                                                        synth_audio_pts += audio_avframe->nb_samples;
+#endif
                                                        avcodec_fill_audio_frame( audio_avframe, codec->channels, codec->sample_fmt,
                                                                (const uint8_t*) p, AUDIO_ENCODE_BUFFER_SIZE, 0 );
                                                        int got_packet = 0;
@@ -1705,9 +1720,11 @@ static void *consumer_thread( void *arg )
                                                        }
 #if LIBAVCODEC_VERSION_MAJOR >= 54
                                                        audio_avframe->nb_samples = FFMAX( samples, audio_input_nb_samples );
+#if LIBAVCODEC_VERSION_MAJOR >= 55
                                                        if ( audio_codec_id == AV_CODEC_ID_VORBIS )
                                                                audio_avframe->pts = synth_audio_pts;
                                                        synth_audio_pts += audio_avframe->nb_samples;
+#endif
                                                        avcodec_fill_audio_frame( audio_avframe, codec->channels, codec->sample_fmt,
                                                                (const uint8_t*) audio_buf_2, AUDIO_ENCODE_BUFFER_SIZE, 0 );
                                                        int got_packet = 0;
@@ -2016,9 +2033,11 @@ static void *consumer_thread( void *arg )
 #if LIBAVCODEC_VERSION_MAJOR >= 54
                                pkt.size = audio_outbuf_size;
                                audio_avframe->nb_samples = FFMAX( samples / channels, audio_input_nb_samples );
+#if LIBAVCODEC_VERSION_MAJOR >= 55
                                if ( audio_codec_id == AV_CODEC_ID_VORBIS )
                                        audio_avframe->pts = synth_audio_pts;
                                synth_audio_pts += audio_avframe->nb_samples;
+#endif
                                avcodec_fill_audio_frame( audio_avframe, c->channels, c->sample_fmt,
                                        (const uint8_t*) p, AUDIO_ENCODE_BUFFER_SIZE, 0 );
                                int got_packet = 0;
@@ -2200,6 +2219,18 @@ on_fatal_error:
                free( full );
                free( cwd );
                remove( "x264_2pass.log.temp" );
+
+               // Recent versions of libavcodec/x264 support passlogfile and need cleanup if specified.
+               if ( !mlt_properties_get( properties, "_logfilename" ) &&
+                     mlt_properties_get( properties, "passlogfile" ) )
+               {
+                       file = mlt_properties_get( properties, "passlogfile" );
+                       remove( file );
+                       full = malloc( strlen( file ) + strlen( ".mbtree" ) + 1 );
+                       sprintf( full, "%s.mbtree", file );
+                       remove( full );
+                       free( full );
+               }
        }
 
        while ( ( frame = mlt_deque_pop_back( queue ) ) )